[asterisk-users] Using Asterisk Only as Voice RecordingSolution.

[Syed Nasruddin]

Thanks Steve,

Sure, although I would have loved to see a pre-config dialplan....:)))).
Thanks for the tip. I think it will help me through.

Best Regards

Syed Nasruddin 



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve
Totaro
Sent: Friday, June 13, 2008 4:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Using Asterisk Only as Voice
RecordingSolution.

Step five: Profit ;-)

I am not going to write your dialplan for you but here is a clue.
http://www.voip-info.org/wiki/view/Asterisk+legacy+integration

Of those various setups, you can extract what you need.

Thanks,
Steve T

On Fri, Jun 13, 2008 at 8:05 AM, Syed Nasruddin <nasruddin at ncel.com.pk>
wrote:
> Dear PaulH,
>
> I have 5 PSTN Lines going into my legacy PBX. There is an active IVR
> present on legacy PBX which the client wants to keep. So what I have
to
> do is:
>
> 1. Install 10 Port i.e 5-FXO/5-FXS card on asterisk machine.
> 2. Insert All those PSTN directly to my 5-Port FXO.
> 3. Take out 5-FXS Port lines and insert them into my legacy PBX.
> 4. Since as I mentioned previously that my client wants to keep its
IVR
> intact on its Legacy system so I will not be handling IVR in my
Asterisk
> Dialplan.
> 5. when the call arrives at asterisk....what should I do?? Should I
> simply call Dial(FXS channel) or something else.
>
> Kindly provide some info regarding Step 5.
>
> Thanks
>
> Syed Nasruddin
>
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Paul
Hales
> Sent: Friday, June 13, 2008 9:38 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Using Asterisk Only as Voice
> RecordingSolution.
>
>
> Basically, you run the phone lines into the asterisk box, then out of
> the Asterisk system into the PABX.
>
> This works reasonably well, and gives you the option to migrate to a
> full asterisk setup in the future.
>
> PaulH
>
>
>
> Syed Nasruddin wrote:
>> Thanks Steve,
>>
>> How I can use it "Asterisk" as Man In The Middle. Since we have to
> keep
>> our Native PBX intact and functioning but only thing it doesn't
handle
>> is Voice Recording. I thought if I can get some Channel Variable or
> some
>> system generated event regarding OFF-HOOK and ON-HOOK condition
> through
>> Asterisk I will easily handle this requirement.
>>
>> It will be a great help if you can elaborate how I can use asterisk
as
>> man-in-the-middle configuration along with my current PBX.
>>
>> Thanks a lot for your prompt response
>>
>> Syed Nasruddin (CISSP)
>>
>> Assistant Manager - Systems
>> National Commodity Exchange Limited
>> 9th Floor, PIC Towers
>> 32-A Lalazar Drive
>> M.T. Khan Road
>> Karachi
>> Phone: 111623623 ext 217
>> Fax: 5611263
>> Web: www.ncel.com.pk
>>
>>
>> -----Original Message-----
>> From: asterisk-users-bounces at lists.digium.com
>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve
>> Totaro
>> Sent: Thursday, June 12, 2008 7:39 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Using Asterisk Only as Voice
>> RecordingSolution.
>>
>> On Thu, Jun 12, 2008 at 11:16 AM, Syed Nasruddin
> <nasruddin at ncel.com.pk>
>> wrote:
>>
>>> HI,
>>>
>>>
>>>
>>> I am using TDM800P Digium Card with Asterisk 1.4.* version. I have
>>>
>> fair
>>
>>> command over Asterisk up till now and have run it in different
>>>
>> scenarios
>>
>>> such as Call Center Solution, PBX solution.
>>>
>>>
>>>
>>> There is a requirement to use Asterisk only as Voice Recording
>>>
>> solution in
>>
>>> following manner:
>>>
>>>
>>>
>>> Physical POT lines before entering into our native PBX will be
>>>
>> splitted and
>>
>>> one of each of those lines will also enter into our Asterisk System.
>>> Once the call is routed by our native PBX and recipient picks up the
>>>
>> phone
>>
>>> (either SIP phone or Analog Phone) I should be able to start
> recording
>>>
>> the
>>
>>> call.
>>> When the call ends, the recording should stop.
>>>
>>>
>>>
>>> Problem being faced by me is this that I am able to catch the call
in
>>>
>> my
>>
>>> diaplan and initialize MixMonitor but since my diaplan never detects
>>> OFF-HOOK/ON_HOOK condition it continue to wait and finally hang-up
>>>
>> while in
>>
>>> actual the call is running through our PBX.
>>>
>>>
>>>
>>> Is there any channel variable or any other mechanism by which I can
>>> accomplish this task? Since i will not be using any Dial() or
similar
>>> application I will be needing some kind of OFF-HOOK trigger/Event in
>>>
>> my
>>
>>> dialplan.
>>>
>>>
>>>
>>> Your help will be highly appreciated.
>>>
>>>
>>>
>>> regards
>>>
>>>
>>>
>>> Syed Nasruddin
>>>
>>>
>>
>> It may not be possible to do this in parallel the way you are trying
>> now.  In series should be a simple task.
>>
>> Just pass the call through Asterisk as the man in the middle, the
>> dialplan will be very simple.
>>
>> Thanks,
>> Steve T
>>
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